The present invention relates generally to conducting conference calls over a computer network. More specifically, the present invention is concerned with conducting conference calls where the number of participants is likely to affect call quality.
With today's interconnected world, Voice over IP (VoIP) has become one of the most rapidly growing technologies in telecommunications. Although VoIP is in general much cheaper than traditional dedicated telephone lines, VoIP as a technology is very resource intensive in terms of the number of data packets that need to be transferred over a network in order to run a single call. This vast amount of data represents a high traffic load on any network and eventually consumes a considerable part of the available bandwidth of the network. Thanks to the increasing processing power and memory availability in modern servers, it has become feasible to encode VoIP data packets using very sophisticated codecs to highly compress the data. This means that it is possible to save on transmission bandwidth without trading off on call quality. However, this has led to the wide availability and much reduced cost of VoIP services. This has attracted many more users than before to use such services and this places extra stress on the data networks being used.
VoIP performance depends on a number of network-related factors (parameters), including available bandwidth, end to-end delay, packet loss and jitter. Variance in these parameters often leads to degradation of VoIP performance and the Quality-of-Experience (QoE) or Quality of Service (QoS) perceived by end users. Moreover, other than network issues, application specific factors like the choice of codec, codec parameters, and jitter buffer sizing also impact QoE. It is important for implementers of VoIP applications to assess QoE as perceived by the end user and take mitigating actions when it degrades to unacceptable levels. Mean Opinion Score (MOS) is the commonly accepted metric to measure the QoE of a call as perceived directly by the end user—it encapsulates the effects of both network and implementation specific issues.
In recent years VoIP has become an extremely important application class, with VoIP clients being very widely used by businesses and individuals. The success achieved by the basic two-party VoIP communications in terms of reliability and the cutting of costs has encouraged the emergence of multi-party VoIP conferencing facilities. Intuitively, it is more difficult to ensure QoE in multi-party sessions since at different times during a session, different people, connecting via different network paths, will be speaking.
The Session Initiation Protocol (SIP), used by the majority of VoIP applications, supports establishment of communications sessions with multiple participants. Nevertheless, VoIP applications have considerable flexibility in how VoIP sessions are realized. Currently, VoIP conferences are implemented through three possible connection topologies: Decentralized, Centralized, and Hybrid. The Centralized model is one of the models commonly used in designing VoIP multi-party conferencing systems. Each endpoint is connected directly to the focus (moderator), and it has no current knowledge of other connections between other endpoints and the focus. Multiple links to the focus are often subjected to different degradation factors. A centralized model is based on a central point of control called a focus or the Multi Control Unit (MCU). The MCU is typically responsible for SIP signaling between all the conference endpoints. Moreover, all the transmitted audio data in the conference call must pass first through the focus to be decoded, mixed (if more than one user is speaking) and finally re-encoded and sent to the rest of the endpoints.
When the number of the conference call participants is small e.g., just a few people, usually the call quality is excellent or the best that could be achieved based on the current system and network capabilities. However, when the number of participants starts to grow, e.g., in the case of educational sessions, all-hands meetings, remote round table meetings, etc., the system and network resources may become overloaded such that it is not feasible to maintain the same call quality. Quality of Service (QoS) would therefore be reduced which could cause call troubles and eventually user dissatisfaction.